similar to: How to run different codecs between the same endpoints on an IAX trunk?

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

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Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

I have some classes in my model which use STI and they work as expected in the console. However, when I try to use them in a migration, I get "uninitialized constant OfficePhone", for example. Why doesn''t the migration environment pick up the class defs? OfficePhone is defined in the model/phone.rb. I put model :phone in the application.rb but my migration just

Two * servers: *a and *b. Outside call comes in *b, and is automatically routed to *a. Someone on a sip phone connected to *a then decides to transfer the call to someone on a sip phone connected to *b. The transfer works. At this point, is *a still in the converstation? Or is * smart enough to see where the data stream is going/coming from? Thanks for any help in advanced, and sorry if

Hey guys sorry for this newbie type post but I'm not getting it really. I see other posts that r asking a simliar question but I'm not understanding. Currently I have working boxA and BoxC. SSH is the only thing accepted. So currenty I'm backing up files on boxC with "rsync -e ssh user@publicIPforB:/home/backup/back_me_up /home/backup". Now I have added boxB in front of

We have a load balancer sending requests to one of X boxes and one of N mongrel processes on that box. Since each mongrel processes is multi-threaded but it has a mutex around the section that calls rails, we end up with several requests queued up waiting when they could have gone to another box with a free process. For example, boxA, and boxB. boxA has mongrels 1 through 10 boxB has

Hi, I have searched the archived, and see this question has come up with some frequency, but the solution by the original poster is never posted or the question is not answered completely. I have also RTFM (rsync and ssh), and I've been trying examples that I have googled. So here it is again. BoxA --internet-- BoxB (accessable only by ssh) -- BoxC I want to backup BoxC to BoxA. I want

I'm trying to get two * boxes to talk.... no matter what variation I try I get No Authority Found and connection refused from 192.168.1.5 I've googled, I've site searched.... to no avail. Here is the server a configs (192.168.1.5): iax.conf [general] port=5036 bandwidth=low disallow=all allow=gsm jitterbuffer=yes tos=lowdelay register => pbx:test@192.168.2.2 [pbx] type=peer

hello list, i need to setup an asterisk system with 5 ISDN trunks. i found C4 cards but they are very expensive. i found that if i use 5 AVM Fritz! cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. and i want, this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

The setup... Transport is via SSH. BoxA : Ubuntu 8.10 BoxB : Mac OS X 10.4.11 If I use rsync v2 to transfer data from BoxA to BoxB everything works fine. However, if I use rsync v3 to transfer the data, the following error is reported : rsync: mkstemp "/path/to/somefile/file.pm.JKQczN" failed: No such file or directory (2) If at all possible,

Hello All, This is probably another one of those completely silly questions that I'm going to hit myself later on, but I have the simplest issue right now but I can't figure out why it's happening: I have a trunk from one * box (box a) to another * box (box b) the call comes in from box a with an extension 2222 which acc. to the peer config in sip.conf is set to use context

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from